[asterisk-ss7] asterisk-ss7 Digest, Vol 42, Issue 14

Matthew Fredrickson creslin at digium.com
Thu Aug 14 13:12:39 CDT 2008


anonymus anonymus wrote:
> Hi all:
> 
> I have a questions,  the strategic plain of Asterisk-SS7 not considered 
> evolution to sigtran? Remenber sigtran is a very cost effcient evolution 
> of traditional links SS7, more and more telco's STP today handle sigtran 
> signaling, and I think is more cheap cards ethernet than cards E1's.

I don't think it's a matter of not considering it, I think we definitely 
need to be able to support sigtran.  However, the importance for doing 
it has not been great, since I have mostly been concerned with making 
sure that the configuration we have right now (MTP2/MTP3/ISUP over 
T1/E1) is solid.

I would think that in the future there will probably be some level of 
sigtran support.  I have looked into it a few times, but again, I do not 
have any equipment that supports it.  Before all else though, it boils 
down to this: there are a lot of things I "could" spend my time on, but 
I've been trying to identify the most effective things to spend my time on.

As of late, I have been thinking the channel driver with MGCP and SS7 
support would be the most effective.  However, that project will 
probably include signalling gateway support as well (which will probably 
include sigtran) so I wouldn't get terribly worried that it will not be 
worked on at some point.

Matthew Fredrickson

> 
> Thanks and regards.
> 
> --- El *jue, 14/8/08, asterisk-ss7-request at lists.digium.com 
> /<asterisk-ss7-request at lists.digium.com>/* escribió:
> 
>     De: asterisk-ss7-request at lists.digium.com
>     <asterisk-ss7-request at lists.digium.com>
>     Asunto: asterisk-ss7 Digest, Vol 42, Issue 14
>     Para: asterisk-ss7 at lists.digium.com
>     Fecha: jueves, 14 agosto, 2008 2:00
> 
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> 
>     Today's Topics:
> 
>        1. Re: ss7 newbie (Christopher (JVM))
>        2. Re: ss7 newbie (Matthew Fredrickson)
>        3. Re: ss7 newbie (olivier taylor)
> 
> 
>     ----------------------------------------------------------------------
> 
>     Message: 1
>     Date: Thu, 14 Aug 2008 11:43:47 -0400
>     From: "Christopher (JVM)" <christopher at jvminfo.com>
>     Subject: Re: [asterisk-ss7] ss7 newbie
>     To: asterisk-ss7 at lists.digium.com
>     Message-ID:
>      <20080814153454.M13601 at jvminfo.com>
>     Content-Type: text/plain;	charset=iso-8859-1
> 
>     As a newbie myself I was wondering if someone could tell me the difference 
>     between the pointcode the adjacent point code and the defaultdpc. I am 
>     trying to obtain these infos from my carrier but he is only giving a single 
>     point code.
> 
>     On Wed, 13 Aug 2008 10:49:26 -0500, Matthew Fredrickson wrote
>     > Sriram wrote:
>     > > Hi
>     > > Thnks for the reply...So say if i want 120 channels over ss7 
>     > > signalling...will the telco give me 4 E1s and 1 separate E1 for ss7 
>     > > signalling or will the ss7 be in one of the 4 E1s itself..?
>     > 
>     > Sometimes.  It depends on what you negotiate with the telco you're 
>     > interconnecting with.  Some of them like to waste an E1 on a single 
>     > signalling link though.  And even stupider ones waste a single E1 on 
>     > two signalling links.  It seems
>      ridiculous to mandate redundant 
>     > signalling links and have them running over the same E1 (Don't ask,
>     >  I had somebody come to me having to deal with that).
>     > 
>     > But in any case, you can negotiate with them how you would like to 
>     > set it up typically.
>     > 
>     > > Do i have to just plug in those 4 E1s directly into the TE420P and 
>     > > install libss7 ? thats it ?
>     > 
>     > Yes :-)  And configure it.
>     > 
>     > > Also please tell me what are the advantages of ss7 over PRI in terms
>     of 
>     > > 1.call handling capacity 2. connectivity 3. reliability ..
>     > 
>     > If you need answers to that question, you probably need to research 
>     > more into what SS7 is.
>     > 
>     > Typically, you have a more reliable link since you can have 
>     > redundant signalling channels.  The protocol also supports a lot 
>     > more bearers per signalling link than you can do in PRI (or
>      even 
>     > NFAS PRI).
>     > 
>     > Matthew Fredrickson
>     > Digium, Inc.
>     > 
>     > _______________________________________________
>     > --Bandwidth and Colocation Provided by http://www.api-digital.com--
>     > 
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>     > To UNSUBSCRIBE or update options visit:
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> 
> 
>     --
>     Christopher Srinivasa
>     VoIP Researcher and Developer / Chercheur et Developpeur VoIP
>     christopher at jvminfo.com
>     JVM INFORMATIQUE CANADA INC.
>     9655-E rue Ignace
>     Brossard, Quebec, Canada J4Y 2P3
>     Tel: 514-951-2981 x2034
>     Alt. Tel: 450-444-1884 x2034
>     Fax: 450-444-1884
> 
> 
> 
> 
>     ------------------------------
> 
>     Message: 2
>     Date: Thu, 14 Aug 2008 10:52:31 -0500
>     From: Matthew Fredrickson <creslin at digium.com>
>     Subject: Re: [asterisk-ss7] ss7 newbie
>     To: asterisk-ss7 at lists.digium.com
>     Message-ID:
>      <48A454BF.8000001 at digium.com>
>     Content-Type: text/plain; charset=ISO-8859-1; format=flowed
> 
>     Christopher (JVM) wrote:
>     > As a newbie myself I was wondering if someone could tell me the difference
> 
>     > between the pointcode the adjacent point code and the defaultdpc. I am 
>     > trying to obtain these infos from my carrier but he is only giving a
>     single 
>     > point code.
> 
>     The adjpointcode is the point code of the entity on the remote end of 
>     the sigchan you are using.
> 
>     The defaultdpc is like the default route.  So if your telco has STPs 
>     between you and the entity that you are supposed to be sending your ISUP 
>     traffic to (basically an additional hop), you set the defaultdpc to the 
>     node that your ISUP traffic should go to.  If there are no STPs (or 
>     additional hops) between you and the point code that you are sending 
>     your traffic to, you should just set the adjpointcode and the defaultdpc
>      
>     to be the same thing.
> 
>     Matthew Fredrickson
> 
>     > 
>     > On Wed, 13 Aug 2008 10:49:26 -0500, Matthew Fredrickson wrote
>     >> Sriram wrote:
>     >>> Hi
>     >>> Thnks for the reply...So say if i want 120 channels over ss7 
>     >>> signalling...will the telco give me 4 E1s and 1 separate E1 for
>     ss7 
>     >>> signalling or will the ss7 be in one of the 4 E1s itself..?
>     >> Sometimes.  It depends on what you negotiate with the telco you're
> 
>     >> interconnecting with.  Some of them like to waste an E1 on a single 
>     >> signalling link though.  And even stupider ones waste a single E1 on 
>     >> two signalling links.  It seems ridiculous to mandate redundant 
>     >> signalling links and have them running over the same E1 (Don't
>     ask,
>     >>  I had somebody come to me having to deal with that).
>     >>
>     >> But in any case, you can negotiate with them how
>      you would like to 
>     >> set it up typically.
>     >>
>     >>> Do i have to just plug in those 4 E1s directly into the TE420P and
> 
>     >>> install libss7 ? thats it ?
>     >> Yes :-)  And configure it.
>     >>
>     >>> Also please tell me what are the advantages of ss7 over PRI in
>     terms of 
>     >>> 1.call handling capacity 2. connectivity 3. reliability ..
>     >> If you need answers to that question, you probably need to research 
>     >> more into what SS7 is.
>     >>
>     >> Typically, you have a more reliable link since you can have 
>     >> redundant signalling channels.  The protocol also supports a lot 
>     >> more bearers per signalling link than you can do in PRI (or even 
>     >> NFAS PRI).
>     >>
>     >> Matthew Fredrickson
>     >> Digium, Inc.
>     >>
>     >> _______________________________________________
>     >> --Bandwidth and
>      Colocation Provided by http://www.api-digital.com--
>     >>
>     >> asterisk-ss7 mailing list
>     >> To UNSUBSCRIBE or update options visit:
>     >>    http://lists.digium.com/mailman/listinfo/asterisk-ss7
>     > 
>     > 
>     > --
>     > Christopher Srinivasa
>     > VoIP Researcher and Developer / Chercheur et Developpeur VoIP
>     > christopher at jvminfo.com
>     > JVM INFORMATIQUE CANADA INC.
>     > 9655-E rue Ignace
>     > Brossard, Quebec, Canada J4Y 2P3
>     > Tel: 514-951-2981 x2034
>     > Alt. Tel: 450-444-1884 x2034
>     > Fax: 450-444-1884
>     > 
>     > 
>     > _______________________________________________
>     > --Bandwidth and Colocation Provided by http://www.api-digital.com--
>     > 
>     > asterisk-ss7 mailing list
>     > To UNSUBSCRIBE or update options visit:
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> 
> 
> 
> 
>     ------------------------------
> 
>     Message: 3
>     Date:
>      Thu, 14 Aug 2008 18:17:12 +0200
>     From: olivier taylor <olivier at sunforyou.be>
>     Subject: Re: [asterisk-ss7] ss7 newbie
>     To: asterisk-ss7 at lists.digium.com
>     Message-ID: <48A45A88.3040407 at sunforyou.be>
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